The present invention relates generally to the field of digital signal processing, and, more particularly, to controlling undesirable bias in equalizers.
The demand for remote access to information sources and data retrieval, as evidenced by the success of services such as the World Wide Web, is a driving force for high-speed network access technologies. The public switched telephone network (PSTN) offers standard voice services over a 4 kHz bandwidth. Traditional analog modem standards generally assume that both ends of a modem communication session have an analog connection to the PSTN. Because data signals are typically converted from digital to analog when transmitted towards the PSTN and then from analog to digital when received from the PSTN, data rates may be limited to 33.6 kbps as defined in the V.34 Transmission Recommendation developed by the International Telecommunications Union (ITU).
The need for an analog modem may be eliminated, however, by using the basic rate interface (BRI) of the Integrated Services Digital Network (ISDN). A BRI offers end-to-end digital connectivity at an aggregate data rate of 160 kbps, which is comprised of two 64 kbps B channels, a 16 kbps D channel, and a separate maintenance channel. ISDN offers comfortable data rates for Internet access, telecommuting, remote education services, and some forms of video conferencing. ISDN deployment, however, has generally been very slow due to the substantial investment required of network providers for new equipment. Because ISDN is not very pervasive in the PSTN, the network providers have typically tariffed ISDN services at relatively high rates, which may be ultimately passed on to the ISDN subscribers. In addition to the high service costs, subscribers must generally purchase or lease network termination equipment to access the ISDN.
While most subscribers do not enjoy end-to-end digital connectivity through the PSTN, the PSTN is nevertheless mostly digital. Typically, the only analog portion of the PSTN is the phone line or local loop that connects a subscriber or client modem (e.g., an individual subscriber in a home, office, or hotel) to the telephone company""s central office (CO). Local telephone companies have been replacing portions of their original analog networks with digital switching equipment. Nevertheless, the connection between the home and the CO has been the slowest to change to digital as discussed in the foregoing with respect to ISDN BRI service. A recent data transmission recommendation issued by the ITU, known as V.90, takes advantage of the digital conversions that have been made in the PSTN. By viewing the PSTN as a digital network, V.90 technology can accelerate data downstream from the Internet or other information source to a subscriber""s computer at data rates of up to 56 kbps, even when the subscriber is connected to the PSTN via an analog local loop.
To understand how the V.90 Recommendation achieves this higher data rate, it may be helpful to briefly review the operation of V.34 analog modems. V.34 modems are generally optimized for a configuration in which both ends of a communication session are connected to the PSTN by analog lines. Even though most of the PSTN is digital, V.34 modems treat the network as if it were entirely analog. Moreover, the V.34 Recommendation assumes that both ends of the communication session suffer impairment due to quantization noise introduced by analog-to-digital converters. That is, the analog signals transmitted from the V.34 modems are sampled at 8000 times per second by a codec upon reaching the PSTN with each sample being represented or quantized by an eight-bit pulse code modulation (PCM) codeword. The codec uses 256, non-uniformly spaced, PCM quantization levels defined according to either the xcexc-law or A-law companding standard (i.e., the ITU G.711 Recommendation).
Because the analog waveforms are continuous and the binary PCM codewords are discrete, the digits that are sent across the PSTN can only approximate the original analog waveform. The difference between the original analog waveform and the reconstructed quantized waveform is called quantization noise, which limits the modem data rate.
While quantization noise may limit a V.34 communication session to 33.6 kbps, it nevertheless affects only analog-to-digital conversions. The V.90 standard relies on the lack of analog-to-digital conversions in the downstream path, outside of the conversion made at the subscriber""s modem, to enable transmission at 56 kbps.
The general environment for which the V.90 standard has been developed is depicted in FIG. 1. An Internet Service provider (ISP) 22 is connected to a subscriber""s computer 24 via a V.90 digital server modem 26, through the PSTN 28 via digital trunks (e.g., T1, E1, or ISDN primary Rate Interface (PRI) connections), through a central office switch 32, and finally through an analog loop to the client""s modem 34. The central office switch 32 is drawn outside of the PSTN 28 to better illustrate the connection of the subscriber""s computer 24 and modem 34 into the PSTN 28. It should be understood that the central office 32 is, in fact, a part of the PSTN 28. The operation of a communication session between the subscriber 24 and an ISP 22 is best described with reference to the more detailed block diagram of FIG. 2.
Transmission from the server modem 26 to the client modem 34 will be described first. The information to be transmitted is first encoded using only the 256 PCM codewords used by the digital switching and transmission equipment in the PSTN 28. These PCM codewords are transmitted towards the PSTN 28 by the PCM transmitter 36 where they are received by a network codec. The PCM data is then transmitted through the PSTN 28 until reaching the central office 32 to which the client modem 34 is connected. Before transmitting the PCM data to the client modem 34, the data is converted from its current form as either xcexc-law or A-law companded PCM codewords to pulse amplitude modulated (PAM) voltages by the codec expander (digital-to-analog (D/A) converter) 38. These PAM voltage levels are processed by a central office hybrid 42 where the unidirectional signal received from the codec expander 38 is transmitted towards the client modem 34 as part of a bidirectional signal. A second hybrid 44 at the subscriber""s analog telephone connection converts the bidirectional signal back into a pair of unidirectional signals. Finally, the analog signal from the hybrid 44 is converted into digital PAM samples by an analog-to-digital (A/D) converter 46, which are received and decoded by the PAM receiver 48. Note that for transmission to succeed effectively at 56 kbps, there must be only a single digital-to-analog conversion and subsequent analog-to-digital conversion between the server modem 26 and the client modem 34. Recall that analog-to-digital conversions in the PSTN 28 may introduce quantization noise, which may limit the data rate as discussed hereinbefore. The A/D converter 46 at the client modem 34, however, may have a higher resolution than the A/D converters used in the analog portion of the PSTN 28 (e.g., 16 bits versus 8 bits), which results in less quantization noise. Moreover, the PAM receiver 48 needs to be in synchronization with the 8 kHz network clock to properly decode the digital PAM samples.
Transmission from the client modem 34 to the server modem 26 follows the V.34 data transmission standard. That is, the client modem 34 includes a V.34 transmitter 52 and a D/A converter 54 that encode and modulate the digital data to be sent using techniques such as quadrature amplitude modulation (QAM). The hybrid 44 converts the unidirectional signal from the digital-to-analog converter 54 into a bidirectional signal that is transmitted to the central office 32. Once the signal is received at the central office 32, the central office hybrid 42 converts the bidirectional signal into a unidirectional signal that is provided to the central office codec. This unidirectional, analog signal is converted into either xcexc-law or A-law companded PCM codewords by the codec compressor (A/D converter) 56, which are then transmitted through the PSTN 28 until reaching the server modem 26. The server modem 26 includes a conventional V.34 receiver 58 for demodulating and decoding the data sent by the V.34 transmitter 52 in the client modem 34. Thus, data is transferred from the client modem 34 to the server modem 26 at data rates of up to 33.6 kbps as provided for in the V.34 standard.
The V.90 standard offers increased data rates (e.g., data rates up to 56 kbps) in the downstream direction from a server to a subscriber or client. Upstream communication still takes place at conventional data rates as provided for in the V.34 standard. Nevertheless, this asymmetry may be particularly well suited for Internet access. For example, when accessing the Internet, high bandwidth is most useful when downloading large text, video, and audio files to a subscriber""s computer. Using V.90, these data transfers can be made at up to 56 kbps. On the other hand, traffic flow from the subscriber to an ISP consists mainly of keystroke and mouse commands, which are readily handled by the conventional rates provided by V.34.
As described above, the digital portion of the PSTN 28 transmits information using eight-bit PCM codewords at a frequency of 8000 Hz. Thus, it would appear that downstream transmission should take place at 64 kbps rather than 56 kbps as defined by the V.90 standard. While 64 kbps is a theoretical maximum, several factors prevent actual transmission rates from reaching this ideal rate. First, even though the problem of quantization error has been substantially eliminated by using PCM encoding and PAM for transmission, additional noise in the network or at the subscriber premises, such as non-linear distortion and crosstalk, may limit the maximum data rate. Furthermore, the xcexc-law or A-law companding techniques do not use uniform PAM voltage levels for defining the PCM codewords. The PCM codewords representing very low levels of sound have PAM voltage levels spaced close together. Noisy transmission facilities may prevent these PAM voltage levels from being distinguished from one another thereby causing loss of data. Accordingly, to provide greater separation between the PAM voltages used for transmission, not all of the 256 PCM codewords are used.
It is generally known that, assuming a convolutional coding scheme, such as trellis coding, is not used, the number of symbols required to transmit a certain data rate is given by Equation 1:
bps=Rs log2 Nsxe2x80x83xe2x80x83EQ. 1
where bps is the data rate in bits per second, Rs is the symbol rate, and Ns is the number of symbols in the signaling alphabet or constellation. To transmit at 56 kbps using a symbol rate of 8000, Equation 1 can be rewritten to solve for the number of symbols required as set forth below in Equation 2:
Ns=256000/8000=128xe2x80x83xe2x80x83EQ. 2
Thus, the 128 most robust codewords of the 256 available PCM codewords are chosen for transmission as part of the V.90 standard.
The V.90 standard, therefore, provides a framework for transmitting data at rates up to 56 kbps provided the network is capable of supporting the higher rates. The most notable requirement is that there can be at most one digital-to-analog conversion and no analog-to-digital conversion in the downstream path in the network. Nevertheless, other digital impairments, such as robbed bit signaling (RBS) and digital mapping through PADs, which results in attenuated signals, may also inhibit transmission at V.90 rates. Communication channels exhibiting non-linear frequency response characteristics are yet another impediment to transmission at the V.90 rates. Moreover, these other factors may limit conventional V.90 performance to less than the 56 kbps theoretical data rate.
In addition to the foregoing factors, errors in demodulating the V.90 signal in the client modem 34 may also affect V.90 performance. The PAM receiver 48 may include an equalizer, such as a decision feedback equalizer (DFE), for demodulating the incoming V.90 signal. During startup procedures for the client modem 34, equalizer training is typically performed in which the serve modem 26 sends a binary signal with a constant amplitude and a sign bit controlled by a scrambler circuit to the client modem 34. As this signal propagates through the digital portion of the network, digital impairments, such as digital attenuation PADs and RBS may alter it. Digital attenuation PADs may be compensated for in the client modem 34 by boosting the gain applied to the incoming signal through, for example, an automatic gain control circuit. Unfortunately, RBS may alter the levels (i.e., PAM signal levels) of some symbols (i.e., PCM codewords) relative to others.
To overcome the effects of RBS during equalizer training, the client modem 34 preferably selects a symbol that is consistent regardless of any RBS that may be used in the digital network. This ideal solution, however, may be very difficult to achieve in practice. For example, it may be impossible to choose a symbol that is unaffected by RBS after attenuation by a digital PAD for all possible digital PADs that may be used.
The digital PSTN 28 transports information using a six symbol framing structure. That is, a frame comprises six data frame intervals with each data frame interval holding a single symbol. Any pattern of altered PAM signal levels caused by RBS will, therefore, repeat every six symbols. Thus, during equalizer training, the training symbol may be consistently biased away from the expected PAM level in one or more of the data frame intervals as a result of RBS. The equalizer will typically attempt to compensate for this bias by updating its filter coefficients, which may introduce an unwanted bias into the filter coefficients and limit the achievable performance for the connection.
Consequently, there exists a need for improvements in modem receivers that may reduce the impact of RBS on equalizer training.
It is an object of the present invention to provide bias control systems, methods, and computer program products that may improve equalizer performance.
It is another object of the present invention to provide bias control systems, methods, and computer program products that may be used to reduce the bias that may be introduced into equalizer filter coefficients due to RBS.
These and other objects, advantages, and features of the present invention may be provided by bias control systems, methods, and computer program products in which an error signal is generated that corresponds to a difference between a reference signal and an equalizer output signal. The error signal is then filtered using a first filter circuit to generate an error signal average. If the absolute value of the error signal does not exceed the error signal average or, alternatively, a suitable threshold proportional to the error signal average, then the error signal is coupled to the equalizer for use in updating the filter coefficients. Furthermore, in accordance with another aspect of the present invention, a second filter circuit may be used to generate an average of selected equalizer output signal samples. If the absolute value of the error signal is greater than the error signal average or, alternatively, a suitable threshold proportional to the error signal average, then the reference signal is updated to correspond to the average of selectee equalizer output signal samples.
Large errors may be interpreted as resulting from inaccurate reference signals or reference levels. The reference signals or reference levels may be inaccurate due to the effects of digital impairments in the network, such as RBS, exhibited in the equalizer output signal. Advantageously, the reference signal or reference level is updated to correspond to average of selected equalizer output signal samples rather than using the error signal to update the equalizer filter coefficients. Conversely, small errors may be interpreted as an indication that the reference signals or reference levels are accurate and do not require additional refinement. In this case, the error signal is used to update the equalizer filter coefficients.
The ITU V.90 Recommendation provides for data transmission via data frames. A data frame comprises six data frame intervals with each interval holding a single symbol. Moreover, the data frame intervals may be exposed to different digital impairments, such as different RBS schemes and/or different PAD attenuation levels. Therefore, in accordance with yet another aspect of the present invention, a magnitude of the reference signal is stored for each of a plurality of data frame intervals.
The average of equalizer output signal samples may be generated by multiplying the magnitude of the reference signal for one of the plurality of data frame intervals by a first weight factor, multiplying the equalizer output signal by a second weight factor, and then adding the results of the two multiplication operations to arrive at a new value for the reference signal. The two weight factors are preferably fractional values whose sum is equal to one. In a preferred embodiment, the first weight factor is set to {fraction (63/64)} and the second weight factor is set to {fraction (1/64)}.
Similarly, the error signal average may be generated by multiplying a previous error signal average by a first weight factor, multiplying the absolute value of the error signal by a second weight factor, and then adding the results of the two multiplication operations. The two weight factors are preferably fractional values whose sum is equal to one. In a preferred embodiment, the first weight factor is set to {fraction (31/32)} and the second weight factor is set to {fraction (1/32)}.
In accordance with still another aspect of the present invention, the error signal may be scaled before it is used to generate the error signal average. The scaling constant or factor may be adjusted so that the error signal average is a suitable threshold for concluding that the error signal is sufficiently large to indicate that the reference signal is inaccurate.
Thus, the bias control systems, methods, and computer program products of the present invention may be used to address the problem of an equalizer in a client modem adjusting its coefficients during the training procedure to adapt to both biased training signal PAM levels (i.e., those affected by digital impairments) and unbiased training signal PAM levels. Such adaptation may result in a compromise set of equalizer coefficients that are biased away from the values that would be attained absent the effects of digital impairments in the network. By using the error signal for coefficient updating only when the error signal is xe2x80x9csmallxe2x80x9d (i e., less than or equal to an error signal average, which is typically based on a scaled version of the error signal), the likelihood of introducing an undesirable bias into the equalizer coefficients may be reduced.